In this paper, the method of synthesizing arbitrary view images that uses 16 cameras on a plane is examined. First of all, we propose the technique for estimating accurate depth information between the cameras and objects. Next, the real arbitrary view images are synthesized by using the 3D shape model estimated by the depth information and the actual images appropriately. The images at arbitrary view points can be generated without errors due to occlusion.
{"title":"Synthesis of Arbitrary View Images Using Depth Estimation Based on Iterative Comparison","authors":"Yasuyuki Haruta, Akira Kubotat, Ryutaro Oi, Takayuki Hamamoto","doi":"10.1109/ISPACS.2006.364721","DOIUrl":"https://doi.org/10.1109/ISPACS.2006.364721","url":null,"abstract":"In this paper, the method of synthesizing arbitrary view images that uses 16 cameras on a plane is examined. First of all, we propose the technique for estimating accurate depth information between the cameras and objects. Next, the real arbitrary view images are synthesized by using the 3D shape model estimated by the depth information and the actual images appropriately. The images at arbitrary view points can be generated without errors due to occlusion.","PeriodicalId":178644,"journal":{"name":"2006 International Symposium on Intelligent Signal Processing and Communications","volume":"79 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126322903","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2006-12-01DOI: 10.1109/ISPACS.2006.364876
R. Banchuin, R. Chaisricharoen, B. Chipipop, B. Sirinaovakul
Commonly known, the gyrator-based OTA simulated floating inductor can be divided into two categories; 3-OTA and 4-OTA structure which perform identically in the ideal phenomena where all OTA's nonidealities i.e. parasitic elements, effect of finite open-loop bandwidth and noise have been neglected. It has been found in R. Banchuin et al. (2005) that the 4-OTA-based floating inductor has better functional and noise performances than its 3-OTA counterpart in the practical phenomena where all of the cited nonidealities included. However, this conclusion has been made based upon the assumption that all OTAs are of the bipolar type. Therefore, due to the rise of the age of CMOS technology; an attempt to find the difference between the 3-CMOS-OTA and 4-CMOS-OTA based floating inductors has been made. Including all of the cited nonidealities, the 4-CMOS-OTA-based floating inductor also has both better functional and noise performances than its 3-CMOS-OTA counterpart. This conclusion strengthens the superiority of the 4-OTA structure over the 3-OTA counterpart since it has been found to be independent of the basis technology and also supports the design guideline proposed in R. Banchuin et al. (2005)
{"title":"In Depth Analysis of The CMOS OTA-Based Floating Inductors","authors":"R. Banchuin, R. Chaisricharoen, B. Chipipop, B. Sirinaovakul","doi":"10.1109/ISPACS.2006.364876","DOIUrl":"https://doi.org/10.1109/ISPACS.2006.364876","url":null,"abstract":"Commonly known, the gyrator-based OTA simulated floating inductor can be divided into two categories; 3-OTA and 4-OTA structure which perform identically in the ideal phenomena where all OTA's nonidealities i.e. parasitic elements, effect of finite open-loop bandwidth and noise have been neglected. It has been found in R. Banchuin et al. (2005) that the 4-OTA-based floating inductor has better functional and noise performances than its 3-OTA counterpart in the practical phenomena where all of the cited nonidealities included. However, this conclusion has been made based upon the assumption that all OTAs are of the bipolar type. Therefore, due to the rise of the age of CMOS technology; an attempt to find the difference between the 3-CMOS-OTA and 4-CMOS-OTA based floating inductors has been made. Including all of the cited nonidealities, the 4-CMOS-OTA-based floating inductor also has both better functional and noise performances than its 3-CMOS-OTA counterpart. This conclusion strengthens the superiority of the 4-OTA structure over the 3-OTA counterpart since it has been found to be independent of the basis technology and also supports the design guideline proposed in R. Banchuin et al. (2005)","PeriodicalId":178644,"journal":{"name":"2006 International Symposium on Intelligent Signal Processing and Communications","volume":"31 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130177853","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Location estimation algorithm for a pedestrian is proposed for the real surveillance system based on color instances with color dynamic images under low illumination, where the proposed color instances consist of color-difference, moving possibility region, and previous detection objects in edge region using time series data. It provides useful detection result for too low illuminated situation. Experimental results for dynamic image taken under low illumination in streets show that detected frames with the proposed algorithm increase by 20% compared with detection result without color instances. The proposed algorithm is under consideration for use in a relatively poor security downtown area in Japan.
{"title":"Instance-based location estimation algorithm for a pedestrian in multiple color dynamic images","authors":"Yutaka Hatakeyama, Akimichi Mitsutat, Kaoru Hirota","doi":"10.1109/ISPACS.2006.364712","DOIUrl":"https://doi.org/10.1109/ISPACS.2006.364712","url":null,"abstract":"Location estimation algorithm for a pedestrian is proposed for the real surveillance system based on color instances with color dynamic images under low illumination, where the proposed color instances consist of color-difference, moving possibility region, and previous detection objects in edge region using time series data. It provides useful detection result for too low illuminated situation. Experimental results for dynamic image taken under low illumination in streets show that detected frames with the proposed algorithm increase by 20% compared with detection result without color instances. The proposed algorithm is under consideration for use in a relatively poor security downtown area in Japan.","PeriodicalId":178644,"journal":{"name":"2006 International Symposium on Intelligent Signal Processing and Communications","volume":"22 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128005058","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2006-12-01DOI: 10.1109/ISPACS.2006.364804
M. Suzuki, Ming Zhang, Haihua Chen, Tingting Teng
This paper proposes a novel design method of criteria for detecting the number of signals superimposed in multichannel time-series. Based on probabilistic properties of difference in maximum log likelihood at infinite SNR, penalty functions in information theoretic criteria are designed by giving specific upper bounds of error probabilities. The proposed design method uses an approximation of probability distribution functions of the difference in maximum log likelihood are approximated. Finally, simulation results are shown to demonstrate flexible criteria for detecting the number of signals can be designed in the case that the number of available samples of observation vectors is small and also large
{"title":"A Design of Information Theoretic Criteria for Detecting the Number of Incoherent Signals","authors":"M. Suzuki, Ming Zhang, Haihua Chen, Tingting Teng","doi":"10.1109/ISPACS.2006.364804","DOIUrl":"https://doi.org/10.1109/ISPACS.2006.364804","url":null,"abstract":"This paper proposes a novel design method of criteria for detecting the number of signals superimposed in multichannel time-series. Based on probabilistic properties of difference in maximum log likelihood at infinite SNR, penalty functions in information theoretic criteria are designed by giving specific upper bounds of error probabilities. The proposed design method uses an approximation of probability distribution functions of the difference in maximum log likelihood are approximated. Finally, simulation results are shown to demonstrate flexible criteria for detecting the number of signals can be designed in the case that the number of available samples of observation vectors is small and also large","PeriodicalId":178644,"journal":{"name":"2006 International Symposium on Intelligent Signal Processing and Communications","volume":"7 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130067402","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2006-12-01DOI: 10.1109/ISPACS.2006.364860
M. Akhaee, S. Ghaemmaghami, N. Khademi
This paper presents a novel approach to audio signals watermarking in the wavelet or the Walsh transform domain. The idea is to embed watermark data in the coefficients of some scales of the transform domain. The overall bit rate of this method is about 90 bps. Due to low computational complexity of the suggested approach, particularly in the Walsh domain, this algorithm can be implemented in real time. Experimental results show robustness of the proposed method in low SNRs and also against some typical attacks, such as MP3 compression, echo, filtering, etc. Subjective evaluation confirms transparency of the watermarked audio signals
{"title":"A Novel Technique for Audio Signals Watermarking in the Wavelet and Walsh Transform Domains","authors":"M. Akhaee, S. Ghaemmaghami, N. Khademi","doi":"10.1109/ISPACS.2006.364860","DOIUrl":"https://doi.org/10.1109/ISPACS.2006.364860","url":null,"abstract":"This paper presents a novel approach to audio signals watermarking in the wavelet or the Walsh transform domain. The idea is to embed watermark data in the coefficients of some scales of the transform domain. The overall bit rate of this method is about 90 bps. Due to low computational complexity of the suggested approach, particularly in the Walsh domain, this algorithm can be implemented in real time. Experimental results show robustness of the proposed method in low SNRs and also against some typical attacks, such as MP3 compression, echo, filtering, etc. Subjective evaluation confirms transparency of the watermarked audio signals","PeriodicalId":178644,"journal":{"name":"2006 International Symposium on Intelligent Signal Processing and Communications","volume":"40 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122356939","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2006-12-01DOI: 10.1109/ISPACS.2006.364902
A. N. Iyer, U. Ofoegbu, R. Yantorno, B. Y. Smolenski
A novel approach to performing speaker clustering in telephone conversations is presented in this paper. The method is based on a simple observation that the distance between populations of feature vectors extracted from different speakers is greater than a preset threshold. This observation is incorporated into the clustering problem by the formulation of a constrained optimization problem. A modified c-means algorithm is designed to solve the optimization problem. Another key aspect in speaker clustering is to determine the number of clusters, which is either assumed or expected as an input in traditional methods. The proposed method does not require such information; instead, the number of clusters is automatically determined from the data. The performance of the proposed algorithm with the Hellinger, Bhattacharyya, Mahalanobis and the generalized likelihood ratio distance measures is evaluated and compared. The approach, employing the Hellinger distance, resulted in an average cluster purity value of 0.85 from experiments performed using the switchboard telephone conversation al speech database. The result indicates a 9% relative improvement in the average cluster purity as compared to the best performing agglomerative clustering system
{"title":"Blind Speaker Clustering","authors":"A. N. Iyer, U. Ofoegbu, R. Yantorno, B. Y. Smolenski","doi":"10.1109/ISPACS.2006.364902","DOIUrl":"https://doi.org/10.1109/ISPACS.2006.364902","url":null,"abstract":"A novel approach to performing speaker clustering in telephone conversations is presented in this paper. The method is based on a simple observation that the distance between populations of feature vectors extracted from different speakers is greater than a preset threshold. This observation is incorporated into the clustering problem by the formulation of a constrained optimization problem. A modified c-means algorithm is designed to solve the optimization problem. Another key aspect in speaker clustering is to determine the number of clusters, which is either assumed or expected as an input in traditional methods. The proposed method does not require such information; instead, the number of clusters is automatically determined from the data. The performance of the proposed algorithm with the Hellinger, Bhattacharyya, Mahalanobis and the generalized likelihood ratio distance measures is evaluated and compared. The approach, employing the Hellinger distance, resulted in an average cluster purity value of 0.85 from experiments performed using the switchboard telephone conversation al speech database. The result indicates a 9% relative improvement in the average cluster purity as compared to the best performing agglomerative clustering system","PeriodicalId":178644,"journal":{"name":"2006 International Symposium on Intelligent Signal Processing and Communications","volume":"36 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126683159","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2006-12-01DOI: 10.1109/ISPACS.2006.364733
P. Tsai, T. Lee, T. Chiueh
This paper presents a fast Fourier transform (FFT) processor suitable for IEEE 802.16e (WiMax) OFDM mode. FFT/IFFT processors are very crucial in OFDM transceivers and they usually consume considerable power as well as occupy large area. The proposed FFT processor combines the pipelined architecture and the memory-based architecture so that it can operate at the sample rate and thus achieve power efficiency. The processor is based on the multipath delay commutator architecture with high-radix arithmetic units and two main memories for input buffering, intermediate storage, and output reordering. A proposed conflict-free memory addressing strategy makes possible continuous-flow FFT processing. Simulation results show that it achieves a 29% saving in power consumption.
{"title":"Power-Efficient Continuous-Flow Memory-Based FFT Processor for WiMax OFDM Mode","authors":"P. Tsai, T. Lee, T. Chiueh","doi":"10.1109/ISPACS.2006.364733","DOIUrl":"https://doi.org/10.1109/ISPACS.2006.364733","url":null,"abstract":"This paper presents a fast Fourier transform (FFT) processor suitable for IEEE 802.16e (WiMax) OFDM mode. FFT/IFFT processors are very crucial in OFDM transceivers and they usually consume considerable power as well as occupy large area. The proposed FFT processor combines the pipelined architecture and the memory-based architecture so that it can operate at the sample rate and thus achieve power efficiency. The processor is based on the multipath delay commutator architecture with high-radix arithmetic units and two main memories for input buffering, intermediate storage, and output reordering. A proposed conflict-free memory addressing strategy makes possible continuous-flow FFT processing. Simulation results show that it achieves a 29% saving in power consumption.","PeriodicalId":178644,"journal":{"name":"2006 International Symposium on Intelligent Signal Processing and Communications","volume":"8 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127143232","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2006-12-01DOI: 10.1109/ISPACS.2006.364854
N. Eiamjumrus, S. Aramvith
Base on the observation that Cauchy distribution provides accurate estimates of rate and distortion characteristics of video sequences, in this paper, we propose a new rate control scheme based on Cauchy based rate-distortion optimization model for the application of H.264 bit allocation. One solution which has been proposed in this paper uses the Langrange multiplier technique as the cost function to find the rate and distortion model subject to the target bit rate constraint resulting in the optimum choice of quantization step sizes. Model parameters are estimated using statistical linear regression analysis. Accordingly we then propose a simple rate control scheme using this new Cauchy rate-distortion model. The target number of bit for each frame is determined according to their buffer status, combined with the number of bits use in the previous frame. The technique proposed has been implemented in H.264 video encoder. Experimental results showed that the proposed rate control algorithm achieves an improvement of average PSNR with smooth video quality compared with the H.264 JM8.6 rate control
{"title":"New rate control Scheme based on Cauchy Rate-Distortion Optimization Model for H.264 Video Coding","authors":"N. Eiamjumrus, S. Aramvith","doi":"10.1109/ISPACS.2006.364854","DOIUrl":"https://doi.org/10.1109/ISPACS.2006.364854","url":null,"abstract":"Base on the observation that Cauchy distribution provides accurate estimates of rate and distortion characteristics of video sequences, in this paper, we propose a new rate control scheme based on Cauchy based rate-distortion optimization model for the application of H.264 bit allocation. One solution which has been proposed in this paper uses the Langrange multiplier technique as the cost function to find the rate and distortion model subject to the target bit rate constraint resulting in the optimum choice of quantization step sizes. Model parameters are estimated using statistical linear regression analysis. Accordingly we then propose a simple rate control scheme using this new Cauchy rate-distortion model. The target number of bit for each frame is determined according to their buffer status, combined with the number of bits use in the previous frame. The technique proposed has been implemented in H.264 video encoder. Experimental results showed that the proposed rate control algorithm achieves an improvement of average PSNR with smooth video quality compared with the H.264 JM8.6 rate control","PeriodicalId":178644,"journal":{"name":"2006 International Symposium on Intelligent Signal Processing and Communications","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125517787","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2006-12-01DOI: 10.1142/S0219467808002952
Jianming Lu, Ling Wang, Yeqiu Li, T. Yahagi
When a signal is embedded in an additive Gaussian noise, its estimation is often done by finding a wavelet basis that concentrates the signal energy in few coefficients and then thresholding the noisy coefficients. However, in many practical problems such as medical X-ray image, astronomical and low-light images, the recorded data is not modeled by Gaussian noise but as the realization of a Poisson process. Multiwavelet is a new development to the body of wavelet theory. Multiwavelet simultaneously offers orthogonality, symmetry and short support which are not possible in scalar 2-channel wavelet systems. After reviewing this recently developed theory, a new theory and algorithm for denoising medical X-ray images using multiwavelet multiple resolution analysis (MRA) are presented and investigated in this paper. The proposed covariance shrink (CS) method is used to threshold wavelet coefficients. The form of thresholds is carefully formulated which is the key to more excellent results obtained in the extensive numerical simulations of medical image denoising compared to conventional methods
{"title":"Noise Removal for Medical X-ray images in Multiwavelet Domain","authors":"Jianming Lu, Ling Wang, Yeqiu Li, T. Yahagi","doi":"10.1142/S0219467808002952","DOIUrl":"https://doi.org/10.1142/S0219467808002952","url":null,"abstract":"When a signal is embedded in an additive Gaussian noise, its estimation is often done by finding a wavelet basis that concentrates the signal energy in few coefficients and then thresholding the noisy coefficients. However, in many practical problems such as medical X-ray image, astronomical and low-light images, the recorded data is not modeled by Gaussian noise but as the realization of a Poisson process. Multiwavelet is a new development to the body of wavelet theory. Multiwavelet simultaneously offers orthogonality, symmetry and short support which are not possible in scalar 2-channel wavelet systems. After reviewing this recently developed theory, a new theory and algorithm for denoising medical X-ray images using multiwavelet multiple resolution analysis (MRA) are presented and investigated in this paper. The proposed covariance shrink (CS) method is used to threshold wavelet coefficients. The form of thresholds is carefully formulated which is the key to more excellent results obtained in the extensive numerical simulations of medical image denoising compared to conventional methods","PeriodicalId":178644,"journal":{"name":"2006 International Symposium on Intelligent Signal Processing and Communications","volume":"29 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124074770","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
Pub Date : 2006-12-01DOI: 10.1109/ISPACS.2006.364706
Dongguo Li, Katsumi Ymashita
Since introduces intercarrier interference (ICI) is the well-known main barrier of upgrading transmission performance for mobile OFDM systems, pilot-aided technique is regarded as the effective solution even some efficient bandwidth has to sacrifice. Some others proposed blind and semi-blind method for the channel estimation utilizes certain underlying statistical properties of the transmitted data make systems becomes complexity. In this paper, an effective channel estimation and equalization method with pilot-free is proposed, which not only can significantly mitigate the ICI and improve the BER performance but also can upgrade the system transmission efficiency, with practicability. According to the Monte Carlo simulations, the empirical results show that our pilot-free method can approach same performance as the known channel results
{"title":"Channel Estimation Based on Pilot-Free Method for Mobile OFDM Systems","authors":"Dongguo Li, Katsumi Ymashita","doi":"10.1109/ISPACS.2006.364706","DOIUrl":"https://doi.org/10.1109/ISPACS.2006.364706","url":null,"abstract":"Since introduces intercarrier interference (ICI) is the well-known main barrier of upgrading transmission performance for mobile OFDM systems, pilot-aided technique is regarded as the effective solution even some efficient bandwidth has to sacrifice. Some others proposed blind and semi-blind method for the channel estimation utilizes certain underlying statistical properties of the transmitted data make systems becomes complexity. In this paper, an effective channel estimation and equalization method with pilot-free is proposed, which not only can significantly mitigate the ICI and improve the BER performance but also can upgrade the system transmission efficiency, with practicability. According to the Monte Carlo simulations, the empirical results show that our pilot-free method can approach same performance as the known channel results","PeriodicalId":178644,"journal":{"name":"2006 International Symposium on Intelligent Signal Processing and Communications","volume":"73 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2006-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127393730","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
京公网安备 11010802042870号
京ICP备2023020795号-1
扫码关注我们
求助内容：
应助结果提醒方式：